In audio coding systems it is common to exploit different properties of different filter banks for different encoding and decoding steps. For example, a modified discrete cosine transform (MDCT) may be used for encoding the waveform of a digital audio signal prior to transmittal from the encoder to the decoder, and a quadrature mirror filter (QMF) bank may be used for high frequency and spatial synthesis of the digital audio signal in the decoder. In such case, the digital audio signal has to be transformed from a first frequency domain associated with a first filter bank or transform to a second domain associated with a second filter bank or transform in the decoder.
There are systems which, in connection to transforming a digital audio signal from one frequency domain to another, sub-sample the digital audio signal in order to reduce the size of the transforms. This is possible for band-limited digital audio signals and reduces the computational complexity. For example, the High-Efficiency Advanced Audio Coding (HE-AAC) codec operates in a dual rate mode in which the transforms are sub-sampled by a factor of two. Another example is given in US2016035329 A1, where sub-sampling of the digital audio signal is used in order to decrease computational complexity. In these systems the factor by which the transforms are sub-sampled is constant, and does hence not adapt to variations in the digital audio signal. There is thus room for improvements.